General Overview of Spatial Impression,

Envelopment, Localization, and Externalization

David Griesinger

Lexicon,Bedford,MA,USA

This paper provides an overview of the current knowledge of the problems of spatial impression, envelopment, and
localization in enclosed spaces. The emphasis will be on small spaces, but the approach draws heavily on concert
acoustic studies. The literature on this subject has been contradictory and controversial. In the last few years a consen-
sus has been developing that envelopment is primarily determined by lateral reflected energy arriving at least 80 ms af-
ter the direct sound. Localization is traditionally linked to the apparent source width (ASW), which has been associated
with the interaural correlation (IACC) of the first 80 ms of the impulse response of a space. Although ASW and IACC
may be useful in concert hall measurement, they do not work well in small spaces. This paper will present two new
measures: the diffuse field transfer function (DFT) and the average interaural time delay (AITD). The DFT is a mea-
sure of envelopment, and is useful both in small and large rooms. The AITD is a measure for "externalization," a sonic
property unique to small rooms. A closely related measure, the net interaural time delay (NITD), is useful in under-
standing localization in small spaces.

1. INTRODUCTION Listening rooms also suffer from a perceptual

In the best concert halls and opera houses low
frequency sounds envelop the listeners,
Although one is aware that the attack of the
kettledrums come from the stage or the pit, the
ring of the drum and the rumble of the bass drum
come from all around the hall. The bass viols
and the cellos have the same property,
particularly when they play pizzicato. One of the
joys of an organ concert is hearing the bass swirl
around the cathedral when a pedal note is held.
When the acoustics produce envelopment music
has a living quality that is highly prized by
conductors and players. When recorded music is
played through loudspeakers, envelopment can
often seem adequate at frequencies above
1000Hz, but poor at lower frequencies. In fact,
many recording engineers seem to be unaware
that low frequency envelopment is either
possible or desirable,
Envelopment at higher frequencies can also play
unexpected tricks. Normally the sound image
from a conventional stereo system stays fixed
between the two loudspeakers. But this is not
always the case. Occasionally sound seem to
surround the listener, even in a non reflective
room.
anomaly that has no obvious counterpart in
performance spaces. Low frequency instruments
in popular music, such as the kick drum and the
bass guitar, are almost always perceived as
coming from inside the head. This perception
does not occur in the concert venue, even when
these instruments are amplified. This "in the
head" localization is unique to recorded music.
It is always perceived as artificial by the author.
In this paper we will use the word
"externalization" to describe this perceptual
property.
Both externalization and envelopment depend
strongly on the recording technique, but they
appear to be independent of each other. In
rooms where low frequency envelopment is
perceptible, low frequency instruments in
classical music are often perceived as external,
while low frequenciesin popular music are often
"in the head".
Both envelopment and externalization are highly
dependent on properties of the room. Years ago
we noticed that it is possible to perceive low
frequency envelopment in some home listening
rooms, and not in others. In [43] we attempted
to studyenvelopmentthroughmeasurements of
localization. We noted that in many listening
rooms it was possible to localize low frequencies

136 AES 15th INTERNATIONAL CONFERENCE

 GENERAL OVERVIEW OF SPATIAL IMPRESSION, ENVELOPMENT, LOCALIZATION, AND EXTERNALIZATION
to a particular loudspeaker, but phantom images
were unstable. Panning low frequencies between
two loudspeakers did not yield the same
positional dependence as is noted at higher
frequencies. The phantom image tended to pull
to the center of the listener's head - and be
judged as closer to the center than it was
intended to be.
In [43] we found that the apparent position of the
low frequency sound could be brought more into
alignment with the high frequency sound if the
separation at low frequencies was increased
electronically. The circuit, dubbed a "spatial
equalizer" has become popular with many
engineers. However the primary virtue of the
spatial equalizer for these engineers turned out
not to be improved localization, but enhanced
envelopment. The spatial equalizer works by
increasing the left minus right (L-R) component
of the sound below 300I-Iz. In most listening
rooms the improvement in localization is subtle,
but the improvement in envelopment is obvious,
In the succeeding years we noticed that this
circuit is completely inaudible in some rooms,
Ironically, in most sound mixing studios the
circuit is inaudible because the antiphase
component of the Iow frequency sound cannot be
heard. These rooms typically have a high degree
of symmetry and carefully control the low
frequency reverberation time. The widespread
use of such rooms for sound mixing has had at
least two undesirable effects. These rooms tend
to make professional engineers unaware that low
frequencies can be enveloping. These rooms
also encourage the use of microphone techniques
that enhance imaging at the expense of
envelopment. For example, recording techniques
that utilize only closely spaced omnidirectional
microphones (such as most binaural techniques)
produce excellent imaging with earphones, but
poor low frequency envelopment with
loudspeakers. If your room does not permit you
to hear envelopment, you will not know what
you aremissing,
The externalization of low frequencies is even
more mysterious. Typical home stereo systems
often externalize low frequencies, whereas
symmetrical listening spaces are the most likely
to sound artificial. These rooms are often
described by their owners as possessing "tight"
low frequency imaging. To my ears the low
frequencies are centered, but they are unlike
anything one W°uld hear in a concert,
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So we have at least three mysteries to untangle.
First, why do some rooms support low frequency
envelopment, and what can be done to provide it
in rooms that do not? Second, why do the kick
drum and the bass guitar almost always end up
banging away inside your head and what can we
do to get them out? Third, why do some
recordings sound enveloping even when you
listen to them through two front loudspeakers in
a relatively non reflective room?
2. ENVELOPMENT IN CONCERT HALLS
The study of envelopment in concert halls has
been marked by conlradictions between common
observation and accepted theory. In [44] we
outline a theoretical framework that resolves
these discrepancies. The framework has the
following major parts:
1. Envelopment at low frequencies is perceived
when the interaural time delay fluctuates at a
rate of between 3Hz and 20Hz. Above
400Hz fluctuations in the interaural intensity
difference (IID) and fluctuations in the lTD
are both important.
Below 400Hz the interaural time delay is the
principle cue for localizing the horizontal
direction (azimuth) of low frequency sounds. In
the absence of reflections, the lTD determines
azimuth with high accuracy - within a few
degrees at frequencies of 500Hz and above.
Below 500Hz the accuracy is proportional to the
frequency, so that localization to +-20 degrees is
still possible in the 63Hz octave band.
Lateral reflected energy causes the lTD - and
thus the perceived azimuth - to shift. When the
sound source is broad-banded, or consists of a
musical tone with vibrato, the shift in lTD
becomes a fluctuation. For sources of both
speech and music the fluctuation is essentially
random (or chaotic) in nature. Fluctuations at
rates slowerthan 3Hz are perceivedas source
motion. Above this frequency they are perceived
as envelopment.
2. Where the sound source consists contains
rapid attacks - such as the start of a speech
phoneme, or the attack of a musical note -
the onset of the sound at the listener is
uncorrupted by reflections. In this case the
ITD during the attack accurately determines
the sound direction, and later fluctuations
137
DAVID GRIESINGER
produce envelopment. Thus it is possible to
have both good localization (a low apparent
source width) and high envelopment at the
same time. At higher frequencies the IID
performs a similar role, and localization is
determined by both IID and ITD.
3. For speech or music that is relatively
transparent to reverberation, the fluctuations
are maximal during the pauses between
phonemes or notes. The loudness of the
reverberation during these pauses
determines the degree of envelopment. The
importance of the reverberation during
pauses is increased by the formation of the
background sound stream. Where the
background stream is easily separated from
the foreground stream, envelopment is -6dB
stronger for a given direct to reverberant
ratio,
4. For thickly orchestrated music where such
pauses are rare, the overall amount of
fluctuation determines the degree of
envelopment. This is the case when the
source is relatively continuous, such as
massed strings, or a pink noise test signal.
5. In carefully controlled loudspeaker
experiments by Morimoto's group in Kobe,
it has been found that reflections that come
fi.ombehind the listener are more
enveloping than reflections that come fxom
the front. They have proposed measuring
envelopment through the front/back ratio,
Statement number 5 indicates that the
envelopment perception does not solely arise
from interaural fluctuations. Where it is possible
to discriminate between fi.ont and rear sound,
rear sound is perceived as more enveloping. We
know from the physics of sound perception that
it is possible to discriminate fi-ontsound fi.om
rear sound in two ways. At low frequencies the
two can be discriminated through small
movements of the listener's head, that cause
predictable changes in the ITDs. Unfortunately
head movements produce no shifts in ITDs when
the sound field is largely diffuse - this
localization cue works only with sound that is
well localized. At high fi.equencies it is possible
to discriminate sound that comes from behind by
notches in the frequency spectrum at about
5kHz. These notches are present when sound
comes from greater than about 150 degrees from
thefront,
138
We conclude that the front/back ratio is primarily
important for sound that includes a substantial
amount of energy above 3kHz. With such sound
sources it is most likely possible to make the
front/back discrimination during the separation
of sound into a foreground and a background
stream. Thus the front/backratio is likely to
contribute to both the continuous form of
envelopment, and its perceptually much more
important relative, the background perception.
The statements given above outline the
relationship between the physical sound field and
the perception of envelopment. Within this
framework there are several unavoidable
difficulties. For example, it is plain from
statements three and four that the perceived
degree of envelopment will depend both on the
loudness ofthesoundsource,andthetypeof
music being played. This effect is easily
observed, and it does not make it easy to develop
reliable measures. Statement number 5 implies
that the spectrum of the source and the spectrum
of the reflections will make a significant
difference to how envelopment is perceived.
For frequencies below 1000Hz the perception of
envelopment depends on maximizing the
fluctuation in the interaural time delay of the
sound field at the listener. For music that is
relatively transparent to reverberation it is the
reverberant component of the perceived sound
that creates envelopment. For more continuous
music, it is the total fluctuation in the lTD that
counts.
For frequencies below 1000Hz in small listening
rooms the reverberation time is usually
sufficiently short that the room itself is unable to
develop significant fluctuations in the ITD in the
pauses between syllables or notes. Thus creating
the perception of envelopment in the background
stream depends on how the room acoustics
interact with the material (particularly the
reverberation) on the recording. The
loudspeakers and the room form a transfer
system, which ideally can transmit the
enveloping properties of the recording to the
listener. Our job is to find a way of measuring
the effectiveness of this transfer system, and then
to find how the transfer can be optimized.
At frequencies above 1000Hz the front/back
ratio will have important implications for our
choiceofloudspeaker positions.
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 GENERAL
OVERVIEW
OFSPATIAL
IMPRESSION,
3. ENVELOPMENT AT HIGH
FREQUENCIES
In the frequency band between 300Hz and
3000Hz envelopment is determined by a
combination of fluctuations in the IID and the
lTD. This fact was discovered independently by
the author and Blauert. Blauert notes in [7] and
[8] that fluctuations in IID have slightly different
perceptual properties than fluctuations in lTD.
He concludes that different brain structures are
involved in the two perceptions. Envelopment at
frequencies below 300Hz is determined almost
entirely by fluctuations in the lTD. The
important virtue of the framework presented in
[44] for the perception of envelopment is that the
concept of interaural fluctuations takes the study
of envelopment out of the realm of
psychoacoustics and into the realm of physics,
We can model the mechanisms that create these
fluctuations, and we can measure them.
Once we have moved the problem into the realm
of physics, we can see that the problem of
envelopment has two parts - the nature of the
envelopment receiver, and the nature of the
sound field surrounding the receiver.
The receiver for envelopment is the human head,
the outer and inner ears, and the brain. The
soundfield is the combination of direct and
reflected energy at the listening position. The
head/ear system is the antenna for the perception
of envelopment. Like other antenna, this system
has directional properties. We can study the
directional dependence of the head/ear system
for single reflections by convolving test signals
with published Head Related Transfer Functions
(HRTFs) such as the ones put on the web by Bill
Gardner. After the convolution we can detect
and measure the interaural fluctuations that result
using the software detectors that will be
described later in this paper,
This work is yet to be done. However several
years ago the author did models of the high
frequency behavior of interaural fluctuations
using a simple head model. We found that the
resulting fluctuations depend on the azimuth of
the reflection. For 1000 Hz the optimum angle
lies in a cone centered on a line drawn through
the listener's head. This cone makes an angle of
about 60 degrees from the front of the head. At
about 1700Hz the cone includes the standard
loudspeaker positions at +- 30 degrees from the
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15th
INTERNATIONAL
CONFERENCE
ENVELOPMENT,
LOCALIZATION,
ANDEXTERNALIZATION
front. This data agrees with subjective
experiments into the angular dependenceof
ASW and spaciousness reported by Ando.
(Ando and Morimoto use the interaurai cross
correlation (IACC) as one of their measures of
the spatial properties of concert hail sound.
Morimoto uses the IACC of the first 80ms of a
binaural impulse response as a measure.
Beranek and Hidaka call this measure (IACCe).
For reasons described extensively in [44], the
author finds the measure often misleading,
particularly in small rooms. However the IACCe
does describe the high frequency angular
dependence of envelopment quite well.)
Using interaural fluctuations it can also be
shown that if there are two sound sources in an
anechoic space, and uncorrelated band-limited
noise is played through each source, the sense of
envelopment will be maximum when both
sources are at the optimal angle for the frequency
band in question. For frequencies below 700Hz
the optimal angle is 90 degrees - the sources
must be at the sides of the listener. As the
frequency rises, the optimum position moves
toward the medial plane.
The angular dependence of envelopment at high
frequencies has important consequences for
sound reproduction systems. A few years ago
the author was playing a stereo tape of applause
through a standard two channel system in a
relatively dead room. He was very surprised to
hear the applause coming from ail around him,
even though the loudspeakers were clearly in
front. Adding a bandpass filter quickly showed
that the perception was a simple result of the
angular dependence of high frequency
envelopment. The 1700Hz band produced a
surrounding perception, even though the
loudspeakers were at +-30 degrees. Because the
applause had significant energy in this band, the
result was enveloping. We conclude that
stereophonic reproduction works in part because
envelopment in a particular frequency band can
give an impression of envelopment in all bands,
and even with a relatively narrow front
loudspeaker pair, envelopment at some
frequencies will be high.
Morimoto's front/back ratio becomes important
when the reflected sound contains significant
energy above 3000Hz. In the author's
experience with concert hails and opera houses it
is rare that there is significant energy at those
139
DAVID GRIESINGER
frequencies coming from the rear. Sound from
the rear of a hall is often spatially diffused, and
has been reflected off many surfaces. Typically
each surface takes a toll on the high frequency
content. In halls where an electronic
enhancement system has been installed it is
possible to experiment with the frequency
spectrum of the reflected energy, and invariably
the sound is better if frequencies above 3000I-Iz
are attenuated. Thus the author is not convinced
that the front/back ratio is an important measure
for concerthall design,
However Morimoto has shown conclusively that
the front/back ratio is important when listening
with loudspeakers, and the author can confirm
the observation. This issue becomes quite
important in the design of surround sound
playback systems. Several of the best sound
mixers for surround sound have noted that they
prefer the rear loudspeakers to be placed +- 150
degrees from the front, and not the more typical
+-110 degrees. At 150 degrees these
loudspeakers are capable of producing the
frequency notches in the HRTFs that indicate
rear sound, and speakers at 110 or 120 degrees
cannot. The author is convinced that
loudspeakers at 150 degrees produce a more
exciting surround sound picture. A loud sound
effect from 150 degrees from the front is much
more exciting than one from 120 degrees,
In addition, one of the most effective uses of a
surround sound system for popular music is the
presentation of a live performance, where the
applause and noise from the audience draws the
listener into the experience. Applause has
substantial high frequency content, and the
front/back ratio becomes important,
However, what is good at high frequencies is not
necessarily good at low frequencies. As we will
see, for optimal envelopment at low frequencies
the surround loudspeakers should be at the sides
of the listening area. We might conclude that a
single pair of surround loudspeakers is not
sufficient - although if we are clever there may
be ways around this problem,
4. ENVELOPMENT AT FREQIJENCIES
BELOW IO00Hz IN SMALL ROOMS -
THE DFT
Envelopment below 1000Hz depends on
interaural fluctuations, andon the ability of
human perception to separate sound into a
140
foreground and a background stream. Where
this separation is possible, it is the spatial
properties of the background stream -
specifically the amount of interaural fluctuations
- that determine the envelopment we perceive.
In a small room there is almost never sufficient
late reflected energy to contribute to the
background perception. The late energy must be
supplied by reverberation in the original
recording. To predict the degree of envelopment
we perceive we must be able to predict the
strength of the interaural fluctuationsduring the
reverberant segments of the recording.
Recording engineers know that for best results
the reverberation in a two channel recording
should be uncorrelated - completely different in
the left and right channels. It is the job of the
loudspeaker/room system to cause the listener to
have adequate interaural fluctuations when this
condition occurs. The loudspeaker/room system
is acting as a transfer system, transferring the
decorrelation in the recording to the listener's
ears. We need a measure for how effectively this
transfer works.
Finding a test signal for measuring
envelopment:
This problem turned out to be much more
difficult than expected. Much of the difficulty
lies is choosing a test signal that will adequately
represent music. As mentioned in the section on
concert hall acoustics, the perception of
envelopment is highly influenced by the
presence of gaps in the music that allow
reverberation to be heard. In this study we will
assume that such a gap has already occurred -
we are trying to model the transfer function of
the reverberation within that gap.
The correlation time of musical signals:
However music has another very interesting
property that is highly relevant to this study.
Music generally consists of notes - segments of
sound that have a recognizable pitch. Although
the notes may be rich in harmonics, the
fundamental frequency is often steady.
If we autocorrelate a musical signal with itself
we will see that over a certain length of time the
autocorrelation function is strongly non-zero.
That is, as long as a note continues there will be
a strong fluctuation in the autocorrelation
function. Ando has published material that
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 GENERAL OVERVIE-WOF SPATIAL IMPRESSION, ENVELOPMENT, LOCALIZATION, AND EXTERNALIZATION
found that the average length of the non-zero
region of the autocorrelation function was related
to the type of music being played, with modern
serial music having a short correlation time, and
romantic symphonies having a long correlation
time. Such a result would be expected from the
average length of the notes in the various
musical styles, as well as the presence in many
types of modern music of percussive sounds with
no well established pitch. Ando discovered that
as the correlation length of a musical example
increased, the ideal reverberation time of a
performance space also increased. We will see
that this result can be predicted from the
properties of the DFT.
When we started working for a measure of
envelopment in rooms we used band limited pink
noise as a test signal. Reference [44] contains
several experiments and observations of the
spatial properties of band limited pink noise,
The correlation length of band limited noise
depends on the bandwidth chosen (and to some
degree on the filter type.) For our first work we
choose to use a bandwidth approximately equal
to the width of a critical band in the human ear.
The results showed that this was not a good
choice. Although the results appeared to be
accurately reflect the envelopment of noise
signals in small rooms, they did not predict the
envelopment of musical signals,
As an experiment, we measured the frequency
width and phase fluctuation in the reverberation
from musical notes of various lengths in a real
concert hall (Boston Symphony Hall.) The
resulting reverberation had high coherence. We
were able to use the reverberation in Boston as a
test signal in calculating the DFT, and we found
that to achieve a similar results with noise we
had to use a filter bandwidth of 1-2Hz. A
disadvantage of such a narrow frequency is that
the DFT in a small room is often highly
frequency dependent, and to find an average
value of envelopment one must separately
calculate the DFT at many different frequencies,
Unfortunately there appears to be no short-cut,
We must calculate our measure using a narrow
band test signal,
The reason we need a narrow band test signal is
obvious in hindsight. Small rooms have
relatively short reverberation times, often less
than 0.5 seconds. The time constant of such a
space (the time it takes the sound to decay by
I/e) is the reverberation time divided by 7, or 70
AES 15th INTERNATIONAL CONFERENCE
milliseconds. If the music (or the reverberation
from the music) has a correlation length that is
significantly longer than this time constant the
room does not generate interaural fluctuations
directly. This means that a single driver in the
room will not produce interaural fluctuations at
the ears of a listener. The room can however
detract from the transfer of fluctuations from
multiple drivers, and this is the effect we wish to
measure.
A detector for envelopment at the listening
position
A detector for envelopment is also a difficult
problem. We had hoped to be able to use a
measure developed for concert halls, the
interaural difference, or IAD. To make a long
story short - it doesn't work. We were in fact
unable to find a proxy measure that could even
duplicate the perceived envelopment in an
anechoic space, let alone the envelopment in a
reflective space. For example, it is obvious that
a single sound source in an anechoic space is
incapable of producing envelopment. Our
measure must show that this is the case. By
similar reasoning, two sources driven by highly
correlated material should also give near-zero
envelopment in an anechoic space, and the
measure should reflect this. The IAD fails on
both counts.
(As luck would have it, the IACC measure does
distinguish between echoic and anechoic spaces.
However - again making a long story short - the
IACC fails to predict observed results at
frequencies below 200Hz.)
We found it was necessary to go back to first
principles. The hypothesis in [44] predicts that
the perception of envelopment at low frequencies
depends on fluctuations in the ITD. There is no
shortcut - to measure envelopment we must
convolve the binaural impulse response from
each sound source with an independent signal,
and then measure the fluctuations in the lTD that
result. In practice this is difficult. Evolution has
had millions of years to perfect methods of
extractinglTD informationfromnoisy signalsat
the eardrums. We had to develop an algorithm
in software that make this extraction. Our
current measure still needs some work, but it
seems to give useful results. Using this
algorithm and a narrow band noise signal as a
test probe we developed our measure for
141
DAVID
GRIESINGER
envelopment in small rooms. We call it the
DFT, or diffuse field transfer function,
The process of finding the diffuse field transfer
function can be summarized:
1. Calculate (or measure) separate binaural
impulse responses for each loudspeaker
position to a particular listener position. A
high sample rate must be chosen to maintain
timing accuracy. In our experiments
176400Hz is an adequate sample rate.
2. Low-pass filter each impulse response and
resample at 11025Hz, and then do it again,
ending with a sample rate of 2756Hz. This
sample rate is adequate for the frequencies
of interest, and low enough that the
convolutions do not take too much time.
3. Create a test signals from independent
filtered noise signals. Various frequencies
and bandwidths can be tried, depending on
the correlation time of the musical signal of
interest,
4. Convolve each binaural impulse response
with a different band filtered noise signal,
and sum the resulting convolutions to derive
the pressure at each ear.
5. Extract the lTD from the two ear signals by
comparing the positive zero-crossing time of
eachcycle,
6. Average the ITDs thus extracted to find the
running average lTD. The averaging
process weights each lTD by the
instantaneous pressure amplitude. In other
words, 1TDs where the amplitudes at the two
ears is high count more strongly in the
average than lTDs where the amplitude is
low.
7. Sum the running average lTD and divide by
the length to find the average lTD and the
apparent azimuth of the sound source.
8. Subtract the average value from the running
average lTD to extract the interaural
fluctuations.
9. Filter the result with a 3Hz to 17Hz
bandpass filter to find the fluctuations that
produce envelopment,
142
10. Measure the strength of these fluctuations
by finding the average absolute value of the
fluctuations. The number which results is
the Diffuse Field Transfer function, or DFT.
11. Measure the DFT as a function of the
receiver position in the room under test.
The most difficult part of this process is building
the ITD detector in software. The detector must
be robust. The signals at the ears are noise
signals - in many places the amplitude is low,
and the zero crossings can be highly confused.
Our detector should use very simple elements -
just timers and filters - to do the job. It must be
very difficult to confuse. The design of this
detector is beyond the scope of this paper.
Persons interested in its design, or in the Matlab
code for the whole DFT measurement apparatus,
should contact the author. In the current version
of the code, it takes about 15 seconds to find the
DFT at a single receiver position, so a 7x7 array
of positions can be calculated in about 12
minutes(ona laptopwitha 150MHzPentium.)
The current code uses noise signals with 10000
samples each. This is about 3.7 seconds at the
sample rate of 2756Hz. We are interested in
random fluctuations in a signal in the 3Hz to
17Hz band - with typical maximum energy
around 5Hz. Needless to say, the accuracy of
such a measurement given a 3.5 second duration
is not high. TheDFTweget fa'omthis
measurement has a semi-random variation of
about 2dB, with occasional values as many as
3dB different from what was expected. Thus the
DFT we present here is somewhat noisy. We
could improve the accuracy by increasing the
length of the noise signals - gaining the usual
improvement by a factor of 1.4 for each doubling
of the signal length.
5. CALIBTATION OF THE DFT MODEL
With a single driver in an anechoic space the
DFT should be zero. In practice, in spite of our
limited length of noise, the values we get are at
least 40dB less than the maximum values with
two drivers. Thus our detector passes this test.
5a. Calibration of the DFT for noise signals
There are two major adjustments to the detector
that we must set as best as we can to emulate the
properties of human hearing. The most
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 GENERAL OVERVIEW OF SPATIAL IMPRESSION, ENVELOPMENT, LOCALIZATION, AND EXTERNALIZATION
important of these is the bandwidth we choose
for the noise signal. When we want to study the
envelopment of noise signals in a room we
would like a noise signal with the bandwidth and
filter shape of a single critical band on the basilar
membrane. In the Matlab code we use a sixth
order elliptical bandpass filter, similar to the
ones in a sound level meter. We need to choose
the bandwidth,
In an effort to calibrate the DFT detector, a series
of experiments on the envelopment of low
frequency noise signals was performed. It is
possible to probe the properties of the human
envelopment detector through experiments with
single lateral reflections, and with multiple
lateral reflections. We are interested in how the
envelopment impression depends on the delay of
single reflections, or the combination of delays
in multiple reflections. The apparatus described
in [44] was used, with continuous band filtered
pink noise as a source. The results were highly
interesting,
First, (with a single subject) we found that the
envelopment from a single lateral reflection
depends on the delay of the reflection. There is
an interference effect. For frequencies in the
63Hz octave band a single lateral reflection at
5.5ms delay produces a very wide and
enveloping sound field, with relatively low
sound pressure. A delay of 13ms produces a
nearly monaural impression, with little or no
envelopment at all. As the frequency rises the
envelopment goes through one more cycle,
becoming first super wide, and then somewhat
less wide. Beyond 20ms all delays sound about
the same.
This interference behavior arises from easily
calculated cancellation between the direct sound
and the reflection. Such interference is not
possible when the delays are greater than the
coherence time of the noise signal, and this
depends on its bandwidth. Thus the properties of
the basilar membrane filters can be studied
through the interference effect. We find that at
least at 63Hz the basilar membrane can be
modeled by an elliptical filter of one octave
width. If the basilar membrane was significantly
sharper than one octave at 63Hz, we would
expect the interference effect to extend to greater
delays. If you use a ½ octave filter in the DFT
detector you find that the interference effect with
a single reflection does extend to higher delays,
A one octave filter at 63Hz seems about right,
AES 15th INTERNATIONAL CONFERENCE
As an aside - we also found to our surprise that
when multiple reflections are used the
envelopment and the DFT depend strongly on
the combination of delays chosen. This
observation has pronounced implications for
concert hall design. It appears that when there
are multiple lateral reflections the delay times of
thereflectionsrelativeto each othermattera lot.
In fact, once a pattern has been set, the delay of
this pattern relative to the direct sound can be
varied with no change in envelopment. Some
patterns are highly enveloping (greater than a
diffuse field) and some are not enveloping at all.
It seems that it is not just the total energy in the
early reflections that is spatially important!
5b. Bandwidth of the test signal for music:
When we wish to calculate the DFF for music
the test signal must have a longer correlation
length than for noise. As mentioned above, we
can usereal reverberationasa test signal,orwe
can emulate the reverberation with a noise signal
of 1-2Hz bandwidth. We will show some results
based on narrow band noise.
5c. Other considerations in the DFT
measurement system:
Another physiological variable in the DFT
detector is the time constant used in the running
ITD filter. Without this filter the ITD detector is
hopelessly inaccurate, so there is a considerable
reason to include it. Here we simply guess. A
time constant of about 50ms seems to work well.
In practice the filter is implemented with a
variabletimeconstant.The TCis 50msfor
strong signals, and rises linearly as the signal
amplitude falls. This amplitude dependence
keeps zero crossings at low amplitudes (which
tend to be very noisy) from affecting the running
average very much.
The 3Hz to 17Hz bandwidth used for the
fluctuations is also a bit of a guess. It is based
on measurements made with amplitude
modulated and phase modulated pure tones. The
bottom line is that this bandwidth gives quite
reasonable results, so it seems a good choice for
now.
The output of the DFT measurement is a number
that represents the average of the absolute value
of the interaural fluctuations. It is expressed in
milliseconds. What is the meaning of this
143
DAVID GRIESINGER
number? How large should it be when the
envelopment is "just right", and is it possible for
it to be too high?
We might think we could calibrate the DFT by
using impulse responses measured in concert
halls. This method has some advantages, but is
probably not what we want to do. The DFr
depends ultimately on the relationship between
the strength of the lateral sound field in the
region of the listener with the medial sound field,
In a true diffuse field the medial signal will
dominate the lateral signal, since the medial field
includes both the front/back direction and the
up/down direction. In a concert hall this
situation is altered by the floor reflection. At
63Hz the floor reflection enhances the lateral
direction by canceling the vertical sound waves,
At 128Hz the opposite happens, with the vertical
sound being enhanced by the reflection. Which
situation is optimal? In any case, why should we
expect that a concert hall - even a very good one
- would be optimal?
Once again experiments were performed to
measure the envelopment using the apparatus of
[44]. We found that below about 200Hz the
most pleasing overall sensation of envelopment
occurs when two uncorrelated noise sources are
on opposite sides of a listener in an anechoic
space. (Above this frequency the envelopment
from such an array seems too wide, and a more
diffuse soundfield is preferred.) This implies
that below 200Hz the optimum value of the DFT
is similar to the values we would measure in a
concert hall at 63Hz, where the floor reflection
enhances the lateral component of the sound,
Our conclusion is that a diffuse field is not
optimal at low frequencies. We can calibrate our
DFr detector by measuring its value when it is
exactly between two noise sources in a simulated
anechoic space. The value we get is frequency
dependent, but for the 63Hz octave band it is
about 0.24ms. In the following paper we will
use this measure to study the envelopment at low
frequencies in listening rooms.
In conclusion we believe the DFT is a useful
model for how human hearing perceives
envelopment when the sound source is band
limited pink noise. It is not clear at this time
whether significantly different results would be
obtained with other source signals,
144
6. lTD AND EXTERNAliZATION
The "in the head"perceptionalsoarisesfrom the
behavior of the ITD. When we are outdoors or
in a large space, a sound source at the side
produces ITDs of about 0.75ms at the listener's
head. We easily perceive fi'om this ITD that the
source is external and at the side. When the
sound source is in the medial plane (directly in
front, overhead, or behind) the lTD is much
lower. We perceive the sound as centered. We
can usually tell if the source is directly in front or
behind us. When this is possible, we perceive
the sound as external.
It is well known that our ability to discriminate
between rear sound and frontal sound at low
fxequencies depends on our ability to move the
head. In natural hearing small movements of the
head produce predictable changes in the lTD.
Small head movements localize and externalize a
source in the medial plane.
As we will see, in a small room the situation is
quite different. For medial sources (or for a
phantom image of a center source) a listener will
experience a low ITD regardless of how the head
is rotated. It is not possible to determine if the
sound is from the front or the back. The
perception becomes internal and artificial - a
perception peculiar to recorded music.
It is not necessary that the sound source be
localized to a particular direction for
externalization to occur. If the lTD changes
randomly - not in synchrony with head
movements - externalization is still successful.
The resulting sound is perceived as external and
enveloping. Our job in this paper is to find a
measure that describes the degree to which low
frequencies are externalized in a particular sound
field - ideally from a modeled or measured
binaural impulse response. We can then use the
measure to find how externalization can be
optimized.
7. BINAURAL IMPULSE RESPONSE AND
lTD
The Fourrier transform of the impulse response
yields both pressure and phase as a function of
frequency. To study the case where there is
more than one driver active at the same time we
find it convenient to first find the impulse
response from each source separately. The phase
of the transform from each source is adjusted if
AES15th
INTERNATIONAL
CONFERENCE
 GENERAL
OVERVIEW
OFSPATIAL
IMPRESSION,
necessary, and then the transforms are added to
find the total pressure and phase. The transform
of a binaural impulse response is found by
separately calculating the transforms for each
ear. At low frequencies we assume the head can
be approximated by two omnidirectional
receivers, separated laterally by 25cm. We do
not auempt to model diffraction around the head,
which is minimal at the frequencies of interest,
The lTD as a function of frequency can be found
by finding the phase difference between the
transforms of the left and right binaural
responses. The phase difference is then
converted to a time difference by dividing by
frequency.
It is useful before going further to examine the
perceptual meaning of the impulse response and
its Fourrier transform. Both are mathematical
concepts. It is not obvious that either should
have any relationship to what we hear with
music. The impulse response is the response of
the room to a pistol shot or small explosion,
These are fortunately rare in everyday life.
Nonetheless there is a considerable body of
literature that attempts to relate musical
perceptions to direct features of the impulse
response.
The transform of the impulse response describes
the steady state response of the room to single
frequency sinusoids. These are plausibly
musical. In a smallrooms after the time it takes
for the sound to travel across the room a few
times the pressure is nearly at the steady state
value. Most musical notes are longer than this,
and often the attacks of low frequency
instruments are slow enough that the room can
be modeled by the steady state condition,
However each frequency in the transform
represents only a single musical note. We would
like to know the response of the room over a
range of notes. We need a measure for average
pressure and average phase, where the average is
over frequency,
8. NORMALIZED AVERAGE PRESSURE
(NAP), AND AVERAGE lTD (AITD)
To find the average pressure we could simply
sum the pressure squared over the range of
interest to find the total pressure in the frequency
band. The summation assumes that each
frequency was present in the steady state - which
is not particularly likely - but this measure of
AES15thINTERNATIONAL
CONFERENCE
ENVELOPMENT,
LOCALIZATION,
ANDEXTERNALIZATION
average pressure is useful nevertheless. To make
the measure closer to what we hear we weight
the sum by -3dB per octave, so each 1/3 octave
band gives an equal contribution to the sum.
We start with a binaural impulse response, either
measured or calculated. That is, a separate
impulse response for the left ear and the right ear
of a dummy head.
IL(t) = impulse response for the left ear
IR(t) = impulse response for the right ear
Sum = sum of the frequency bins over the
frequencies of interest
2. FFTL(f) = ffi(IL(t))
3. FFTR(f) = fft(IR(t))
7. Average Pressure ----sqrt(sum((magnitude
FFTL(0^2)/f)
This average pressure contains a frequency
factor, which we would like to eliminate
when we plot pressure as a function of
frequency. We can define a normalized average
pressure:
8. Normalized Average Pressure = Average
Pressure / ( sum (l/sqrt(f))
For Iow frequencies we assume that the
magnitude of FFrL is approximately equal to the
magnitude of FFrR.
We suggested above that the ITD at a particular
frequency could be found by comparing the
phase of the left and right parts of a binaural
response. The problem is that this phase
difference is independent of the pressure at that
frequency. It is possible - in fact it is nearly
always the case - that the frequencies where the
phase difference is the largest are the ones for
which the pressure is the lowest. We are not
likely to take much notice of a large lTD if the
pressure is low. Unlike a lateral figure of eight
microphone, which gives a maximum output at
the nulls of a lateral standing wave, we cannot
determine the ITD of a sound we cannot hear. If
we stand at such a null, we hear nothing.
It would be possible to develop a model that
determines lTD the same way the ear does, by
first separating the sound into critical bands, and
then finding the timing of zero crossings for each
ear. See [44.] As we will see later, such a model
is quite expensive computationally. Since we
145
DAVID GRIESINGER
want to calculate the ArrD at many frequencies
and at many points in a room, there is a strong
incentive to develop an efficient model,
To make an average lTD that is closer to what
we actually hear we weight the ITD with the
pressure, lTDs with Iow pressure are given low
weighting, and ITDs with high pressure are
weighted more strongly. We then normalize the
resulting sum by dividing by the average
pressure ox)er the same frequency range.
Given this description the method of finding the
AITD follows directly:
5. Phase_angle(f) = angle(FFTL)-angle(FFTR)
6. liD(f) = Phase_angle(f)/(2*pi*f)
To find the AITD we multiply the absolute value
of lTD(f) by the absolute value of the pressure,
and sum over frequency. We also apply a -3dB
per octaveweighting to give equal weight to all
bands. The sum is then normalized by dividing
by the weighted sum of the absolute pressure
overthe samefrequencyrange,
6. AITD=
sum(magnitude(FFTL(f)) *abs(ITD(f))/(sqrt(
f))/(average pressure)
It is also possible to define a NET lTD (NITD) -
one which preserves the sign of the ITD in the
sum. The NITD is always less than the AITD,
but it indicates our ability to determine the
direction of a sound source, not just its
externalization.
7. NITD =
sum(magnitude(FFTL(f))*ITD(f)/(sqrt(f))/(a
verage pressure)
Note that for simplicity we use only the
magnitude of FFTL(f) to represent the pressure
at both ears. We could average the magnitudes
of FFTL and FFTR, but there seems to be no
reason to do so in practice,
9. AITD AND NITD - WHAT DO THEY
MEAN?
The NITD is a measure that expresses the
apparent direction Of a sound source in a given
frequency band. Not every frequency within the
band will localize to the same degree. In fact,
due to standing waves some frequencies may
localize to the wrong direction. The N1TD
146
averages out these differences, giving us a lower
value than we would get using individual
frequencies. It does reflect the average
localization of a narrow band noise source. We
are interested in externalization of the sound, and
externalization can take place even in the
absence of correct localization. Thus for
externalization the average absolute value of the
ITD, the A1TD is a better measure.
At low frequencies in a free field - with no
reflections - AITD and NliD are equal, and
independent of frequency. Their value depends
on the angle between the listener and the source,
and can be easily calculated.
If source_angle is the angle between the source
and a line drawn between the listener's ears
(listener facing forward):
Lateral AITD = 0.75ms*cos(source_angle)
os
._°'s1
<2o.4.
_..J 02,
0,
2.54.5 6.5 5.5
8.510.5
12.5
lateral
position-
ft 14.5 17.5
position-ft
front-back
Figure 1: The Lateral AITD in the center of a
17'x23' anechoic space with a driver in the
upper left comer (at. 1',. 1'). Where the value is
high, the sound source will be localized to the
side of the listener.
We can also define a Medial AITD - by simply
rotating the listener's head by 90 degrees in
azimuth and calculating the AITD again. In a
free field - assumingthe source angle has not
changed:
AES 15th INTERNATIONAL CONFERENCE
 GENERAL OVERVIEW OF SPATIAL IMPRESSION, ENVELOPMENT, LOCALIZATION, AND EXTERNALIZATION

At present it is not clear how close the AITD

o.6,
04
g 0.2.
o,
2.5
456585
10.5
_25
lateralposition-ft14.517.5
Figure 2: The Medial AITD in the center of a
17'x23' anechoic space with a driver in the
upper left corner at. 1',. 1' Note that where the
lateral AITD is low, the medial AITD is high.
It makes sense to define a Total AITD, which is
simply the RMS sum of the lateral and the
medial A1TD. Since cos^2 + sin^2 is one, in a
free field. Total AITD = 0.75ms.
08
.__
0.75
<_07
_.055
0.6:
2.5
4565
6510.5
12.5
lateral position- ft 14.5 17.5
Figure 3: The Total AITD (the RMS sum of the
lateral and medial AITDs) in the same space as
figures 2 and 3. Note that the driver can be
accurately localized everywhere, as we would
expect in an anechoic environment. The
differences from 0.75ms are due to sampling
errorsin the model.
The total AlTD is a measure of how easily a
sound source can be localized. In a free field it
has the constant value of 0.75, which means the
source can be localized to its true direction with
full accuracy. In a reflective room the total
AITD is almost always less than this value, as
standing waves tend to reduce the lTD at the
listener. Where the total A1TD is significantly
less than 0.75 the sound source will be "in the
head".
must be to 0.75ms for complete externalization.
In our informal experience a value of 0.3ms or
lower will almost always sound internal. Values
of0.4msorsocansometimes beinternaland
sometimesbe external.Clarifyingthisquestion
willrequirefurtherexperiments.Howeverthe
AITDis usefulasa measure.Wewanttofind
5.5
strategiesof speakerplacementandrecording
techniquethat bringthe A1TDas closeas
front-backposition-_ possible to the free field value. Since in most
casesthe listener doesnot move the head very
much, at low frequencies we are mostly
interested in the lateral AITD.
Models that use these measures to study the
properties of listening rooms will be presented in
the next paper.
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1. Ando, Y. and Singh, P.K. and Kurihara,
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2. Ando, Y. and Kurihara, K., Nonlinear
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3. Barron, M., Spatial Impression due to
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AES 15th INTERNATIONAL CONFERENCE 147

DAVID GRIESINGER
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12. Bradley, J.S., Contemporary approaches
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20. Griesinger, D. Measures of Spatial
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21. Griesinger, D., IALF - Binaural
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29. Griesinger, D., 'Optimum reverberant
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June 1995
30. Hartman, W., Localization of a sound
source in a room. Proceedings of the 8th
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31. Hidaka, T. and Beranek, L. and Okano,
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35. Kreiger, A., Nachhallzeitverlangerung
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36. Morimoto, M. and Maekawa, Z.,
Effects of Low Frequency Components on
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37. Morimoto, M. and Posselt, C.,
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40. Schroeder, M.R. and Gottlieb, D. and
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 GENERAL
OVERVIEW
OFSPATIAL
IMPRESSION,
ENVELOPMENT,
LOCALIZATION,
ANDEXTERNALIZATION

concert halls: Correlation of subjective 43. Griesinger, D. "Spaciousness and

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41. Schubert, P., Die Wahrnehmbarkeit
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721-731

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INTERNATIONAL
CONFERENCE 149